From f6a4080ff56cd2ccdaa3e5bbf16dc172ae838e95 Mon Sep 17 00:00:00 2001
From: Brick <[EMAIL REDACTED]>
Date: Thu, 24 Aug 2023 10:47:37 +0100
Subject: [PATCH] audio_resampleLoss: Add support for multiple channels
---
test/testautomation_audio.c | 38 +++++++++++++++++++++++++------------
1 file changed, 26 insertions(+), 12 deletions(-)
diff --git a/test/testautomation_audio.c b/test/testautomation_audio.c
index fa33f1069e68..6b35a5c11451 100644
--- a/test/testautomation_audio.c
+++ b/test/testautomation_audio.c
@@ -802,18 +802,22 @@ static int audio_resampleLoss(void *arg)
};
int spec_idx = 0;
+ int min_channels = 1;
+ int max_channels = 1 /*8*/;
+ int num_channels = min_channels;
- for (spec_idx = 0; test_specs[spec_idx].time > 0; ++spec_idx) {
+ for (spec_idx = 0; test_specs[spec_idx].time > 0; ++num_channels) {
const struct test_spec_t *spec = &test_specs[spec_idx];
const int frames_in = spec->time * spec->rate_in;
const int frames_target = spec->time * spec->rate_out;
- const int len_in = frames_in * (int)sizeof(float);
- const int len_target = frames_target * (int)sizeof(float);
+ const int len_in = (frames_in * num_channels) * (int)sizeof(float);
+ const int len_target = (frames_target * num_channels) * (int)sizeof(float);
SDL_AudioSpec tmpspec1, tmpspec2;
Uint64 tick_beg = 0;
Uint64 tick_end = 0;
int i = 0;
+ int j = 0;
int ret = 0;
SDL_AudioStream *stream = NULL;
float *buf_in = NULL;
@@ -823,15 +827,20 @@ static int audio_resampleLoss(void *arg)
double sum_squared_error = 0;
double sum_squared_value = 0;
double signal_to_noise = 0;
+
+ if (num_channels > max_channels) {
+ num_channels = 1;
+ ++spec_idx;
+ }
SDLTest_AssertPass("Test resampling of %i s %i Hz %f phase sine wave from sampling rate of %i Hz to %i Hz",
spec->time, spec->freq, spec->phase, spec->rate_in, spec->rate_out);
tmpspec1.format = SDL_AUDIO_F32;
- tmpspec1.channels = 1;
+ tmpspec1.channels = num_channels;
tmpspec1.freq = spec->rate_in;
tmpspec2.format = SDL_AUDIO_F32;
- tmpspec2.channels = 1;
+ tmpspec2.channels = num_channels;
tmpspec2.freq = spec->rate_out;
stream = SDL_CreateAudioStream(&tmpspec1, &tmpspec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(SDL_AUDIO_F32, 1, %i, SDL_AUDIO_F32, 1, %i)", spec->rate_in, spec->rate_out);
@@ -848,7 +857,10 @@ static int audio_resampleLoss(void *arg)
}
for (i = 0; i < frames_in; ++i) {
- *(buf_in + i) = (float)sine_wave_sample(i, spec->rate_in, spec->freq, spec->phase);
+ float f = (float)sine_wave_sample(i, spec->rate_in, spec->freq, spec->phase);
+ for (j = 0; j < num_channels; ++j) {
+ *(buf_in + (i * num_channels) + j) = f;
+ }
}
tick_beg = SDL_GetPerformanceCounter();
@@ -890,13 +902,15 @@ static int audio_resampleLoss(void *arg)
tick_end = SDL_GetPerformanceCounter();
SDLTest_Log("Resampling used %f seconds.", ((double)(tick_end - tick_beg)) / SDL_GetPerformanceFrequency());
- for (i = 0; i < len_out / (int)sizeof(float); ++i) {
- const float output = *(buf_out + i);
+ for (i = 0; i < frames_target; ++i) {
const double target = sine_wave_sample(i, spec->rate_out, spec->freq, spec->phase);
- const double error = SDL_fabs(target - output);
- max_error = SDL_max(max_error, error);
- sum_squared_error += error * error;
- sum_squared_value += target * target;
+ for (j = 0; j < num_channels; ++j) {
+ const float output = *(buf_out + (i * num_channels) + j);
+ const double error = SDL_fabs(target - output);
+ max_error = SDL_max(max_error, error);
+ sum_squared_error += error * error;
+ sum_squared_value += target * target;
+ }
}
SDL_free(buf_out);
signal_to_noise = 10 * SDL_log10(sum_squared_value / sum_squared_error); /* decibel */